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[Incoming Call] Mediation Server responds PRACK with 481

Question
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Hi everybody,
the connection from the Mediation Server to the PSTN is realized through a SIP Trunk. In this case, the mediation server is directly connected with the SBC by the provider. in a Test scenario a PSTN subscriber calls the ocs user (MOC). After the SBC has sent a PRACK request to the mediation server, the server responds with 481 and then followed with a 408 Request Time out. The PSTN caller hears a reject tone after a while. In my opinion, the host part (ProviderDomain) in To header field could be a problem for the Mediation Server because he knows only the IP address of the SBC as the next hop but he does not know the domain of the provider. Perhaps it would make sense that the host part in To header field the IP address or the domain of Mediation server (MyDomain) contains. Or is my suspicion wrong and the problem lies elsewhere? May be the cause is as the RFC 3262/Section 3 it describes, and what will that mean in this case? See SIP Trace.
PS: Outgoing Calls are working properly
OCS 2007 R2
OCS User: +331234567
PSTN User: +331234568
SBC -> OCS
----------------------------------------
Jul 9 15:44:04.761 SBC:Port sent to Mediation:5060
INVITE sip:+331234567@Mediation SIP/2.0
Via: SIP/2.0/TCP SBC:5060;branch=z9hG4bKekvubh104oa1ln671.1
To: <sip:+331234567@ProviderDomain;user=phone>
From: "01234568" <sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
Call-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
CSeq: 1 INVITE
Max-Forwards: 68
Contact: <sip:01234568@SBC:5060;transport=tcp>
Date: Fri, 9 Jul 2010 15:44:04 GMT
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: 100rel
P-Asserted-Identity: <sip:01234568@ProviderDomain>
Accept: application/sdp, application/isup, application/xml, application/dtmf-relay
Content-Type: application/sdp
Content-Length: 236
v=0
o=- 0 59079 IN IP4 SBC
s=IMSS
c=IN SBC
t=0 0
m=audio 56215 RTP/AVP 8 101 18 106
a=rtpmap:101 G726-32/8000
a=rtpmap:106 telephone-event/8000
a=sendrecv
a=ptime:20
a=sqn:0
a=cdsc: 1 image udptl t38OCS answers with 100 Trying
----------------------------------------OCS -> SBC
----------------------------------------
Jul 9 15:44:04.782 On SBC:Port received from Mediation:5060
SIP/2.0 183 Session Progress
FROM: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
TO: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
CSEQ: 1 INVITE
CALL-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
VIA: SIP/2.0/TCP SBC:5060;branch=z9hG4bKekvubh104oa1ln671.1
CONTACT: <sip:med.MyDomain:5060;transport=Tcp;maddr=IP-Mediation>
CONTENT-LENGTH: 260
CONTENT-TYPE: application/sdp; charset=utf-8
ALLOW: UPDATE
REQUIRE: 100rel
SERVER: RTCC/3.5.0.0 MediationServer
Rseq: 1
v=0
o=- 80 1 IN IP4 Mediation
s=session
c=IN IP4 Mediation
b=CT:1000
t=0 0
m=audio 63941 RTP/AVP 8 106
c=IN IP4 Mediation
a=rtcp:63941
a=label:Audio
a=rtpmap:8 PCMA/8000
a=rtpmap:106 telephone-event/8000
a=fmtp:106 0-16
a=ptime:20SBC -> OCS
----------------------------------------
Jul 9 15:44:04.788 On SBC:Port sent to Mediation:5060
PRACK sip:med.MyDomain:5060;maddr=IP-Mediation;transport=Tcp SIP/2.0
Via: SIP/2.0/TCP SBC:5060;branch=z9hG4bKf460eq10507ovp2m1.1
To: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
From: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
Call-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
CSeq: 2 PRACK
Max-Forwards: 68
RAck: 1 1 INVITE
Content-Length: 0Ocs -> SBC
----------------------------------------
Jul 9 15:44:04.790 On SBC:Port received from Mediation:5060
SIP/2.0 481 Call Leg/Transaction Does Not Exist
From: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
To: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
CSEQ: 2 PRACK
CALL-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
VIA: SIP/2.0/TCP SBC:5060;branch=z9hG4bKf460eq10507ovp2m1.1
CONTENT-LENGTH: 0
SERVER: RTCC/3.5.0.0 MediationServerOcs -> SBC
----------------------------------------
Jul 9 15:44:38.233 On SBC:Port received from Mediation:5060
SIP/2.0 408 Request Timeout
From: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
To: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
CSEQ: 1 INVITE
CALL-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
VIA: SIP/2.0/TCP SBC:5060;branch=z9hG4bKekvubh104oa1ln671.1
CONTENT-LENGTH: 0
SERVER: RTCC/3.5.0.0 MediationServerThanks in advance!
Friday, July 16, 2010 1:55 PM
Answers
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Hi Randy.
First thanks for your answer.
I have spoken with ISP and they said that they will make a SIP manipulation on the SBC. They have replaced the host part in the To header field with the IP address (Gateway Listening IP address) of the mediation server and the Incoming calls from PSTN work now.
Thanks again for your help!
- Marked as answer by Reyane Monday, July 26, 2010 1:48 PM
Monday, July 26, 2010 1:48 PM
All replies
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Is your SIP trunk provider listed on the OIP page? The reason I ask, is that it certifies compatability.
If not, you can still probably get it to work with some help from your provider.
Can you confirm from your provider the Audio codec they are using for your trunk? I think I saw G726 in one of the traces. Should be G711
If you have the flexibility to, I would suggest looking into a provider on the OIP page, it will eliminate all of the headaches here with a non certified provider.
http://technet.microsoft.com/en-us/office/bb735838.aspx
Thanks.
Randy Wintle | MCTS: UC Voice Specialization | Winxnet IncFriday, July 23, 2010 7:27 PM -
Hi Randy.
First thanks for your answer.
I have spoken with ISP and they said that they will make a SIP manipulation on the SBC. They have replaced the host part in the To header field with the IP address (Gateway Listening IP address) of the mediation server and the Incoming calls from PSTN work now.
Thanks again for your help!
- Marked as answer by Reyane Monday, July 26, 2010 1:48 PM
Monday, July 26, 2010 1:48 PM